2 * ALSA input and output
3 * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
4 * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
6 * This file is part of Libav.
8 * Libav is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
13 * Libav is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
25 * ALSA input and output: input
26 * @author Luca Abeni ( lucabe72 email it )
27 * @author Benoit Fouet ( benoit fouet free fr )
28 * @author Nicolas George ( nicolas george normalesup org )
30 * This avdevice decoder allows to capture audio from an ALSA (Advanced
31 * Linux Sound Architecture) device.
33 * The filename parameter is the name of an ALSA PCM device capable of
34 * capture, for example "default" or "plughw:1"; see the ALSA documentation
35 * for naming conventions. The empty string is equivalent to "default".
37 * The capture period is set to the lower value available for the device,
38 * which gives a low latency suitable for real-time capture.
40 * The PTS are an Unix time in microsecond.
42 * Due to a bug in the ALSA library
43 * (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
44 * decoder does not work with certain ALSA plugins, especially the dsnoop
48 #include <alsa/asoundlib.h>
49 #include "libavformat/avformat.h"
50 #include "libavutil/opt.h"
52 #include "alsa-audio.h"
54 static av_cold int audio_read_header(AVFormatContext *s1,
55 AVFormatParameters *ap)
57 AlsaData *s = s1->priv_data;
60 enum CodecID codec_id;
61 snd_pcm_sw_params_t *sw_params;
63 #if FF_API_FORMAT_PARAMETERS
64 if (ap->sample_rate > 0)
65 s->sample_rate = ap->sample_rate;
68 s->channels = ap->channels;
71 st = av_new_stream(s1, 0);
73 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
75 return AVERROR(ENOMEM);
77 codec_id = s1->audio_codec_id;
79 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
85 if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
86 av_log(s1, AV_LOG_WARNING,
87 "capture with some ALSA plugins, especially dsnoop, "
90 ret = snd_pcm_sw_params_malloc(&sw_params);
92 av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
97 snd_pcm_sw_params_current(s->h, sw_params);
98 snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
100 ret = snd_pcm_sw_params(s->h, sw_params);
101 snd_pcm_sw_params_free(sw_params);
103 av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
108 /* take real parameters */
109 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
110 st->codec->codec_id = codec_id;
111 st->codec->sample_rate = s->sample_rate;
112 st->codec->channels = s->channels;
113 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
122 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
124 AlsaData *s = s1->priv_data;
125 AVStream *st = s1->streams[0];
127 snd_htimestamp_t timestamp;
128 snd_pcm_uframes_t ts_delay;
130 if (av_new_packet(pkt, s->period_size) < 0) {
134 while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
135 if (res == -EAGAIN) {
138 return AVERROR(EAGAIN);
140 if (ff_alsa_xrun_recover(s1, res) < 0) {
141 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
149 snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
151 pkt->pts = timestamp.tv_sec * 1000000LL
152 + (timestamp.tv_nsec * st->codec->sample_rate
153 - ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
154 / (st->codec->sample_rate * 1000LL);
156 pkt->size = res * s->frame_size;
161 static const AVOption options[] = {
162 { "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
163 { "channels", "", offsetof(AlsaData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
167 static const AVClass alsa_demuxer_class = {
168 .class_name = "ALSA demuxer",
169 .item_name = av_default_item_name,
171 .version = LIBAVUTIL_VERSION_INT,
174 AVInputFormat ff_alsa_demuxer = {
176 NULL_IF_CONFIG_SMALL("ALSA audio input"),
182 .flags = AVFMT_NOFILE,
183 .priv_class = &alsa_demuxer_class,